Merge branch 'master' into emacs-team

1 files changed, 137 insertions, 0 deletions

diff --git a/gnu/packages/patches/webrtc-for-telegram-desktop-unbundle-libsrtp.patch b/gnu/packages/patches/webrtc-for-telegram-desktop-unbundle-libsrtp.patch
new file mode 100644
index 0000000000..486f0b1aa5
--- /dev/null
+++ b/gnu/packages/patches/webrtc-for-telegram-desktop-unbundle-libsrtp.patch
@@ -0,0 +1,137 @@
+From 62672f3756ecf218252098211d78c13369ab6d28 Mon Sep 17 00:00:00 2001
+From: Nicholas Guriev <nicholas@guriev.su>
+Date: Thu, 4 May 2023 16:21:09 +0300
+Subject: [PATCH] Unbundle libSRTP
+
+Avoid private symbols and link against system-wide libSRTP. The excluded code
+in SrtpSession looks unreachable from the call integration in Telegram Desktop.
+---
+ CMakeLists.txt          |  3 +++
+ cmake/libsrtp.cmake     | 13 +++++++++++++
+ src/pc/external_hmac.cc |  1 -
+ src/pc/external_hmac.h  |  9 ++++++---
+ src/pc/srtp_session.cc  | 16 ++++++++++++++--
+ 5 files changed, 36 insertions(+), 6 deletions(-)
+
+diff --git a/CMakeLists.txt b/CMakeLists.txt
+index af7d24c21..66bec8fdf 100644
+--- a/CMakeLists.txt
++++ b/CMakeLists.txt
+@@ -2647,6 +2647,9 @@ if (TG_OWT_USE_PROTOBUF)
+     list(APPEND export_targets proto)
+ endif()
+ 
++if (LIBSRTP_FOUND)
++    target_compile_definitions(tg_owt PRIVATE HAVE_LIBSRTP)
++endif()
+ if (NOT absl_FOUND)
+     list(APPEND export_targets libabsl)
+ endif()
+diff --git a/cmake/libsrtp.cmake b/cmake/libsrtp.cmake
+index 5124312d2..01f051606 100644
+--- a/cmake/libsrtp.cmake
++++ b/cmake/libsrtp.cmake
+@@ -1,3 +1,16 @@
++find_package(PkgConfig REQUIRED)
++pkg_check_modules(LIBSRTP libsrtp2)
++
++if (LIBSRTP_FOUND)
++    add_library(libsrtp INTERFACE EXCLUDE_FROM_ALL)
++    add_library(tg_owt::libsrtp ALIAS libsrtp)
++
++    target_include_directories(libsrtp INTERFACE ${LIBSRTP_INCLUDE_DIRS} ${LIBSRTP_CFLAGS_OTHER})
++    target_link_libraries(libsrtp INTERFACE ${LIBSRTP_LINK_LIBRARIES} ${LIBSRTP_LDFLAGS_OTHER})
++
++    return()
++endif()
++
+ add_library(libsrtp OBJECT EXCLUDE_FROM_ALL)
+ init_target(libsrtp)
+ add_library(tg_owt::libsrtp ALIAS libsrtp)
+diff --git a/src/pc/external_hmac.cc b/src/pc/external_hmac.cc
+index 27b5d0e5a..222f5d9ae 100644
+--- a/src/pc/external_hmac.cc
++++ b/src/pc/external_hmac.cc
+@@ -15,7 +15,6 @@
+ 
+ #include "rtc_base/logging.h"
+ #include "rtc_base/zero_memory.h"
+-#include "third_party/libsrtp/include/srtp.h"
+ 
+ // Begin test case 0 */
+ static const uint8_t kExternalHmacTestCase0Key[20] = {
+diff --git a/src/pc/external_hmac.h b/src/pc/external_hmac.h
+index c5071fc19..8fdc2f1a7 100644
+--- a/src/pc/external_hmac.h
++++ b/src/pc/external_hmac.h
+@@ -30,9 +30,12 @@
+ 
+ #include <stdint.h>
+ 
+-#include "third_party/libsrtp/crypto/include/crypto_types.h"
+-#include "third_party/libsrtp/include/srtp.h"
+-#include "third_party/libsrtp/include/srtp_priv.h"
++#ifdef HAVE_LIBSRTP
++# include <srtp2/auth.h>
++# include <srtp2/srtp.h>
++#else
++# include "srtp_priv.h"
++#endif
+ 
+ #define EXTERNAL_HMAC_SHA1 SRTP_HMAC_SHA1 + 1
+ #define HMAC_KEY_LENGTH 20
+diff --git a/src/pc/srtp_session.cc b/src/pc/srtp_session.cc
+index 7d1aaf2d6..7b5a789b0 100644
+--- a/src/pc/srtp_session.cc
++++ b/src/pc/srtp_session.cc
+@@ -30,8 +30,12 @@
+ #include "rtc_base/thread_annotations.h"
+ #include "rtc_base/time_utils.h"
+ #include "system_wrappers/include/metrics.h"
+-#include "third_party/libsrtp/include/srtp.h"
+-#include "third_party/libsrtp/include/srtp_priv.h"
++
++#ifdef HAVE_LIBSRTP
++# include <srtp2/srtp.h>
++#else
++# include "srtp_priv.h"
++#endif
+ 
+ namespace cricket {
+ 
+@@ -290,6 +294,9 @@ bool SrtpSession::UnprotectRtcp(void* p, int in_len, int* out_len) {
+ bool SrtpSession::GetRtpAuthParams(uint8_t** key, int* key_len, int* tag_len) {
+   RTC_DCHECK(thread_checker_.IsCurrent());
+   RTC_DCHECK(IsExternalAuthActive());
++#ifdef HAVE_LIBSRTP
++  return false;
++#else
+   if (!IsExternalAuthActive()) {
+     return false;
+   }
+@@ -313,6 +320,7 @@ bool SrtpSession::GetRtpAuthParams(uint8_t** key, int* key_len, int* tag_len) {
+   *key_len = external_hmac->key_length;
+   *tag_len = rtp_auth_tag_len_;
+   return true;
++#endif
+ }
+ 
+ int SrtpSession::GetSrtpOverhead() const {
+@@ -336,6 +344,9 @@ bool SrtpSession::GetSendStreamPacketIndex(void* p,
+                                            int in_len,
+                                            int64_t* index) {
+   RTC_DCHECK(thread_checker_.IsCurrent());
++#ifdef HAVE_LIBSRTP
++  return false;
++#else
+   srtp_hdr_t* hdr = reinterpret_cast<srtp_hdr_t*>(p);
+   srtp_stream_ctx_t* stream = srtp_get_stream(session_, hdr->ssrc);
+   if (!stream) {
+@@ -346,6 +357,7 @@ bool SrtpSession::GetSendStreamPacketIndex(void* p,
+   *index = static_cast<int64_t>(rtc::NetworkToHost64(
+       srtp_rdbx_get_packet_index(&stream->rtp_rdbx) << 16));
+   return true;
++#endif
+ }
+ 
+ bool SrtpSession::DoSetKey(int type,